IP networks generally provide an excellent infrastructure for geographically distributing components of a telecommunication system. The underlying IP network is optimal for transmission for control signaling, and, when bandwidth is available, can provide an acceptable Quality of Service (or QoS) or Grade of Service (or GOS) for voice communications. When insufficient network resources are available for voice communications or one or more IP network components are down, voice communications can be adversely impacted.
For example, assume an enterprise network having first and second network regions, with each network region being served by a different media gateway and survivable media server. The primary media server controls the media gateways in a first operational mode. When the WAN is available, a subscriber in the first network region can dial any other subscriber in the first and second network regions simply by dialing that subscriber's extension. In the event of a network failure that causes the primary server to lose control of both of the survivable media servers, the media gateway in the first network region will register with the first survivable media server, and the second media gateway in the second network region will register with the second survivable media server. This is the second operational mode. As a result, a subscriber in one network region is unable to dial directly the extension of a subscriber in another network region. Even though each survivable media server is aware of all of the endpoints in the enterprise network, each media server controls only the endpoints in its respective network region.
A number of techniques have been attempted to address these issues.
In one technique, when a system has multiple communication gateways controlled by a single controller and the private switching facilities inter-connecting these gateways failed, users can “dial-out” on a public network trunk using the public address (or the Direct Inward Dial or DID number) of the destination party. A calling subscriber can dial a PSTN access code, followed by a complete public network number to reach the called subscriber. By way of illustration, instead of dialing a five-digit extension (83594) to reach a telephone in another branch office, a subscriber must dial 9-303-538-3594 during a network failure. This approach requires manual intervention by the user first to recognize that a problem exists, second to determine how to circumvent it, and third to dial the DID number. If the destination party to be reached does not have a public number, he or she cannot be reached directly by the alternate network. In particular, subscribers without a DID number cannot be reached by dialing PSTN numbers without an intervening auto-attendant. Moreover, though some subscribers may be permitted to make and receive extension-dialed calls, they may have restrictions that prevent them from placing or receiving PSTN calls. Finally, no feature transparency is available since calls appear as simple incoming and/or outgoing PSTN calls.
Another technique for managing IP bandwidth usage includes call admission control in which the number of calls across the Wide Area Network or WAN or the bandwidth available for voice calls is limited. Call admission control can result in the call being denied and being forwarded to the callee's voice mail server (if accessible), thereby causing caller frustration.
In yet another technique known as PSTN Fallback™ of Avaya Inc., a call is forced to the PSTN when an IP trunk connection experiences an unacceptable QoS or GOS. With reference to FIG. 1, a multi-enterprise architecture is depicted, each enterprise 100 and 104 having a separate, independent, and active or primary media servers 112 and 116 with resident call controller functionality. Each enterprise also includes a plurality of digital stations 120 and 124, a plurality of IP or Internet Protocol stations 128 and 132, a gateway 136 and 140 and a Local Area Network or LAN 144 and 148. The media servers 112 and 116 are independent in that one media server in one enterprise is generally unaware of the subscriber configuration information, such as extensions, of the other enterprise's subscribers. The gateways 136 and 140 are interconnected by the Public Switched Telephone Network or PSTN 148 and Wide Area Network or WAN 152. When a call is to be placed over the WAN 152, the originating call controller determines the currently measured network delay and packet loss. When either measured variable reaches a predetermined threshold, the call controller automatically takes the idle IP trunk ports out-of-service, i.e., it busies out the ports. The ports remain out-of-service until the measurements return to the low threshold. No new calls are allowed over the IP trunk. Normal or conventional call routing over the PSTN 148 is used for access to the next preference in the route pattern.
In a further technique known as Separation of Bearer and Signaling™ (SBS) of Avaya Inc., the signaling channel for a call is routed over the WAN 152 while the bearer channel is routed over the PSTN 148. The signaling channel in SBS includes SBS call-control signaling and QSIG private-networking protocol information. SBS associates the signaling and bearer channels at the SBS originating and terminating nodes so that they appear to the end users to be a normal, non-separated call. The use of the WAN for signaling traffic and the PSTN for voice bearer traffic addresses a customer need for using small amounts of bandwidth in the IP WAN for signaling traffic, with the voice bearer portion of the call being sent over inexpensive PSTN facilities. Like PSTN Fallback, SBS™ is used in multi-enterprise calls with each enterprise having separate, independent, and active media servers.
PSTN Fallback™ and SBS™ address architectures where there exist multiple, separate system implementations inter-connected by a traditional inter-switch trunking protocol; in other words, they permit inter-connection only of peer-to-peer systems. With the move to larger, single-server IP WAN-connected media gateway distributed systems, there is no longer a need for IP trunks and SBS. Using trunk group administration to limit bandwidth between media servers is not required nor is PSTN Fallback™ when the number of calls exceeds the administered IP trunk member limit. There is no need to embed an intelligent signaling interface between servers over IP WAN facilities given that the system has only a single active or primary server and that all calls across the system appear to be station-to-station calls.
Another technique known as the Survivable Remote Site Telephony™ (SRS Telephony) by Cisco Systems, Inc., involves a primary server (such as CallManager™ by Cisco Systems, Inc.) controlling a plurality of interconnected subnetworks. Each subnetwork includes an IP telephony router and media gateway and is connected to other subnetworks by a WAN and the PSTN. In the event of a WAN link failure resulting in a loss of control by the primary server, SRS Telephony automatically detects the network failure and initiates a process to intelligently auto-configure the router to provide call processing redundancy for the IP phones in that network subnetwork. Link failure is detected by the IP telephones when they are no longer receiving keepalive packets from the primary server. In response, each of the IP telephones registers with the router, which queries the telephone about its configuration and then auto-configures itself. The SRS Telephony software, which is resident in the IP telephony router, is automatically activated and builds a local database of all IP telephones attached to it. When the WAN link is restored, the IP telephones detect keepalive packets from the primary server and revert to it for primary call setup and processing. This configuration, however, is only a partial solution. It is applicable only to IP phones and not to other types of communication devices, such as digital phones. Although IP telephones in each impacted subnetwork are able to call one another using extension dialing (which is typically five or fewer digits) by virtue of the call processing functionality of the local IP telephony router, they are unable to use extension dialing to call IP telephones in other subnetworks of the enterprise network. To make such calls, IP telephone users must still dial the full PSTN number (which is typically seven or more digits). Moreover, automatic feature transparency is not provided in the SRS Telephony product.
There is a need, particularly in a single-server system, for a call control system that manages IP bandwidth usage effectively, particularly during high traffic periods and/or provides an alternate communication path in the event of problems with the WAN.